1. Field of the Invention
The present invention relates to an audio signal coding and decoding device used for repairing the cutting of a sound in, for example, a potable compact disk player.
2. Description of the Prior Art
In a portable compact disk player (hereinafter referred to as a CD player), a sound may, in some cases, be cut off in an audio signal read from a compact disk (hereinafter referred to as a CD) due to flaws and dirt, and disturbances such as vibration. Therefore, the following device has been already developed as a device for preventing a sound from being cut off in the CD player. Specifically, digital audio signals recorded on the CD are read out at a speed in excess of an ordinary reading speed and are stored in a buffer memory, and the digital audio signals are outputted at the ordinary reading speed in the order stored in the buffer memory. When a sound is cut off in the audio signal read out from the CD due to the vibration or the like, writing to the buffer memory is interrupted, and a portion where the sound is cut off is detected to perform repair processing of the cutting of the sound. If time required to detect the portion where the sound is cut off and perform the repair processing of the cutting of the sound is within time corresponding to the capacity of the buffer memory, the continuity of the audio signals outputted from the buffer memory can be held.
If an attempt to directly write the audio signal read from the CD to the buffer memory is made, the buffer memory must have a large capacity. Generally in order to reduce the capacity of the buffer memory, therefore, the audio signal read from the CD is compressed by a coding device, a signal obtained by the compression is written to the buffer memory, and a signal outputted from the buffer memory is extended by a decoding device. A sound cutting preventing reproduction mode for compressing an audio signal to make reproduction while repairing the cutting of a sound is generally provided separately from a normal reproduction mode. Accordingly, in the sound cutting preventing reproduction mode, a certain degree of degradation of the sound quality allowed.
Meanwhile, the audio signal recorded on the CD is 16-bit pulse code modulation (PCM) data sampled at a frequency of 44.1 kHz per channel (ch), and the transfer rate thereof is 705.6 kbps/ch. Examples of audio signal coding systems already suggested include an ATRAC (Adaptive Transform Acoustic Coding) system (transfer rate=about 140 kbps/ch) by SONY Corporation and a PASC (Precision Adaptive Subband Coding) system (transfer rate=192 kbps/ch) by Phillips Corporation. The systems are for obtaining a reproduced sound high in quality by dividing an audio signal into several bands and assigning the number of bits utilizing hearing characteristics to each of the bands to code the audio signal.
In addition to the above described systems, an ATC (Adaptive Transform Coding) system, an ADPCM (Adaptive Differential Pulse Code Modulation) system and the like have been generally known as audio signal coding systems. Particularly, the ADPCM system is the simplest coding system. In this system, it is found that when the amount of information of not less than a predetermined amount (5 bits) is assigned to a prediction error signal, a reproduced sound which is by no means inferior to the original sound is obtained if an ordinary music source is reproduced.
FIG. 1 shows a conventional coding device using an ADPCM system. A differential circuit 20 finds a difference d.sub.n (a prediction error signal) between a 16-bit digital audio signal x.sub.n and a predicted signal y.sub.n. Specifically, the differential circuit 20 finds the prediction error signal d.sub.n on the basis of the following expression (1): EQU d.sub.n .times.x.sub.n -y.sub.n ( 1)
The prediction error signal d.sub.n is sent to an encoder 30. The encoder 30 codes the prediction error signal d.sub.n on the basis of quantizer step-size .DELTA..sub.n to find a code L.sub.n. Specifically, the encoder 30 finds the code L.sub.n on the basis of the following expression (2): EQU L.sub.n = d.sub.n /.DELTA..sub.n ! (2)
Here !is Gauss' notation. Specifically, x! indicates the largest of integers which do not exceed a real number x.
This code L.sub.n is sent to a quantizer 40. The quantizer 40 quantizes the prediction error signal d.sub.n using the code L.sub.n to find a quantized value q.sub.n on the basis of the quantizer step-size .DELTA..sub.n. Specifically, the quantizer 40 finds the quantized value q.sub.n on the basis of the following expression (3): EQU q.sub.n =(L.sub.n +0.5).DELTA..sub.n ( 3)
This quantized value q.sub.n is sent to an adder 50. The adder 50 finds, on the basis of a predicted signal y.sub.n corresponding to a value obtained by sampling the audio signal this time x.sub.n and the quantized value q.sub.n, a predicted signal y.sub.n+1 corresponding to a value obtained by sampling the audio signal the next time x.sub.n+1. Specifically, the adder 50 finds the predicted signal y.sub.n+1 corresponding to the value obtained by sampling the audio signal the next time x.sub.n+1 on the basis of the following express ion (4): EQU y.sub.n+1 =y.sub.n +q.sub.n ( 4)
In this case, an initial value y.sub.0 of the predicted signal is 0.
The quantizer step-size .DELTA..sub.n is updated by a quantizer step-size determining circuit 60. The quantizer step-size determining circuit 60 comprises a multiplier 113, a judging circuit 111, and a coefficient memory 112. A function M (L.sub.n) corresponding to the code L.sub.n is stored in the coefficient memory 112. In the present embodiment, the code L.sub.n is expressed by four bits. Specifically, the most significant bit (MSB) of the code L.sub.n indicates whether the code L.sub.n is positive or negative. The MSB is "1" when the code L.sub.n is negative. Consequently, the code L.sub.n is an integer from -8 to +7.
This quantizer step-size .DELTA..sub.n is inputted to the multiplier 113. The output L.sub.n of the encoder 30 is inputted to the judging circuit 111 . The judging circuit 111 reads out the function M (L.sub.n) corresponding to the inputted code L.sub.n from the coefficient memory 112 and sends the same to the multiplier 113. The multiplier 113 multiplies the inputted quantizer step-size .DELTA..sub.n by the function M (L.sub.n), to find quantizer step-size .DELTA..sub.n+1 corresponding to the value obtained by sampling the audio signal the next time X.sub.n+1. Specifically, the multiplier 113 finds the quantizer step-size .DELTA..sub.n+1 corresponding to the value obtained by sampling the audio signal the next time X.sub.n+1 on the basis of the following expression (5): EQU .DELTA..sub.n+1 =.DELTA..sub.n XM(L.sub.n) (5)
The relationship between the code L.sub.n and the function M (L.sub.n) and the relationship between the code L.sub.n and the quantizer step-size .DELTA..sub.n+1 are shown in Table 1.
TABLE 1 ______________________________________ L.sub.n M (L.sub.n) .DELTA..sub.n+1 ______________________________________ 0 -1 0.9 0.9.DELTA..sub.n 1 -2 0.9 0.9.DELTA..sub.n 2 -3 0.9 0.9.DELTA..sub.n 3 -4 0.9 0.9.DELTA..sub.n 4 -5 1.2 1.2.DELTA..sub.n 5 -6 1.6 1.6.DELTA..sub.n 6 -7 2.0 2.0.DELTA..sub.n 7 -8 2.4 2.4.DELTA..sub.n ______________________________________
Meanwhile, if an attempt to improve the quality of a reproduced sound is made, processing for compressing an audio signal becomes complicated. In each of the above described coding systems such as the PASC system, the ATRAC system and the ATC system, an analysis is conducted in a frequency region. Accordingly, processing becomes complicated, so that the hardware and software scales of the coding device become larger, to increase the cost thereof. Further, also in the ADPCM system, one division (the foregoing expression (2)) and two multiplications (the foregoing expressions (3) and (5)) are carried out in coding one sample. One of the multiplications is a multiplication by a fixed coefficient having a decimal point. Accordingly, processing becomes complicated, so that the hardware and software scales of the coding device become larger, to increase the cost thereof.
In the conventional ADPCM system, the predicted signal y.sub.n+1 corresponding to the value obtained by sampling the audio signal the next time X.sub.n+1 is found by adding the quantized value q.sub.n to the predicted signal y.sub.n corresponding to the value obtained by sampling the audio signal this time x.sub.n as represented by the foregoing expression (4). In a case where the predicted signal is found on the basis of the foregoing expression (4), however, even if the amount of information in excess of a predetermined amount is assigned as the number of quantization bits, quantization noise is increased.
One example of a coding and decoding device employing the ADPCM system is disclosed in Japanese Patent Publication No. 5926/1988. In this coding and decoding device, a difference (a prediction error signal) between an input audio signal and a predicted signal is found by a subtracter. The prediction error signal found by the subtracter is sent to an adaptive quantizer. In the adaptive quantizer, the prediction error signal is divided by a normalizing factor adaptively updated, to be coded. A code found by the adaptive quantizer is sent to an adaptive inverted quantizer having a decoder and a multiplier. In the adaptive inverted quantizer, the code is decoded by the decoder and then, an output of the decoder is multiplied by the normalizing factor by the multiplier. Consequently, a decoded value is obtained. The decoded value is sent to an adaptive predictor comprising an adder and a filter. In the adaptive predictor, this predicted signal is added to the decoded value by the adder and then, a predetermined operation is performed by the filter, to produce the succeeding predicted signal. A coefficient of the filter is adaptively corrected on the basis of the output of the decoder in the adaptive inverted quantizer.
The coding and decoding device described in the above described prior art document has the advantages of resisting errors in a transmission path or a recording medium, being superior in band compression characteristics and being small in scale. In this coding and decoding device, however, when the variation between samples of the input audio signal is large, a prediction error is increased and the normalizing factor (quantizer step-size) is increased, resulting in an increased quantization error.